H323: protocol, gateways, ports, configuration and application

The H323 protocol standard is gradually being phased out, particularly for phones. A few years ago, many phone manufacturers abandoned the trend of creating devices that could use both IP telephony protocols. Today, SIP is mainly used. However, many users have an H323 device and need to know how to operate and set up. The standard still offers better network management and better hardware compatibility. The difference between the two protocols is decreasing with each new version, despite much debate on this issue in the industry.

Briefly about IP-telephony

Video over the Internet is a technology that allows the transmission of voice and data on the same network based on the IP protocol. The term converged networks or IS convergence is often used, which implies a broader concept of integrating all communications: voice, data and video.

This technology has been on the market since the late 1990s, but only recently has it become widespread through improvement and standardizationspeech quality control systems (QoS) and universalization of the Internet service.

An IP telephony system is a set of elements that are properly integrated and used by VoIP-based companies. The main elements of this system are: IP PBX, IP gateway and various IP phones with H323 protocol.

Convergence of business communications - audio, data and video. The IP network is a current trend and brings important benefits to users:

1. Save on calls.
2. Simplify the communications infrastructure.
3. Control optimization.
4. Unification of the telephony system between facilities.
5. Mobility/accessibility for the user.

A multi-service converged network must be properly designed and managed, and therefore aspects such as control over reliability, security, and quality of service (QoS) are necessary to ensure an optimally functioning system.

Two main video communication standards

SIP and H323 protocols are considered the main videoconferencing standards. Various organizations are considering audio and video signaling over IP with different approaches. The International Telecommunication Union (ITU) established H323 as the first multimedia communications protocol over IP. SIP is the IETF (Internet Engineering Task Force) approach for transmitting audio and video over IP.

H323 protocol is an umbrella format that provides a well-defined system architecture, including call control and multimedia. WhileH323 uses telecom approach to voice/video over IP, SIP uses internet approach.

The objects that make up the network: gateways, conference terminals and bridges, and the gatekeeper. The architecture is peer-to-peer and supports communication between users without a centralized controlling entity. The H323 call information is written in binary code with a specific set of translations for each code. This was done to reduce the transmission size and conserve bandwidth. New codes must be agreed upon between the parties prior to being called. The standard may be updated, but any additions to the standard require backward compatibility with the existing one.

Operational Protocol H323

The International Telecommunication Union defined the H323 standard to provide audiovisual sessions over network packets for voice over IP and for videoconferencing on the same basis. It relays messages, offers directory services, terminal access control, resource consumption control, and handles call authorization, and can also route signaling. The standard has access to other networks, performing the functions of data conversion and signal broadcasting.

Key Features:

1. Guarantees Quality of Service (QoS).
2. Does not depend on network topology.
3. Supports gateways and uses more than one channel at the same time: voice, video, data.
4. Allows companies to add functionality to implementrequired interaction features.

Main system components

The H323 implements as basic components: terminals, gateways for connections to PSTN/IN resources, guardians (GateKeepers) for access control, registration and bandwidth, MCU (multi-conference) and control units. Description of the H323 protocol and its components:

1. Terminals - equipment used by users can be implemented either in software using a computer or in hardware - physically.
2. Guardians (GateKeepers) - are the center of every VoIP organization and the equivalent of private branches or UATS (Private Branch eXchange). They usually promote with software.
3. Gateways - connected by switching to the public telephone network, acting transparently to the user.
4. Multipoint VUs - are responsible for managing the conference.

A common standard has been propagated across the wide area network (WAN). It consists of many terminals that it manages. For example, multiple LANs separated by routers.

The H323 protocol stack works on top of the transport and network layers. If the core network is IP based, then H.225.0 audio, video, and RAS packets use the UDP (H.245) protocol for data and control. H.225.0 packet call signaling is transported using reliable TCP - Transmission Control Protocol.

Technologies usingprotocol

An example of businesses using the H323 is an IP telephony company that is implementing an Asterisk PBX with cost billing for phone calls to landlines and mobile phones. The user can perform forwarding to a mobile phone, transfer voice to a phone, make calls, record calls, report calls to the PBX. Asterisk can be implemented on the Linux platform, so there will be no software licensing costs. Benefits of VoIP PBX:

1. Supports SIP, IAX, H323, MGCP, SKINNY and more protocols.
2. Supported codecs: ADPCM, G.711 (A-Law & -Law), G.722, G.723.1 (end-to-end), G.726, G.729 (if commercial license purchased), GSM, iLBC.

IP ports and protocols used by various vendors of H323 devices.

Port	Type	Description	H323 customer	Primer H323	Lifesize Cloud Client	Skype for Business Client
80	Static TCP	HTTP web interface	No		No	No
389	Static TCP	LDAP	No
443	Static TCP	HTTPS & Port Tunneling	No
443	Static TCP	Edgewater / Polycom VBP Access Server	No
443	Static TCP	Providing, ICON He alth Check			No
443	Static TCP	Streaming			No
443	Static TCP	Desktop / Mobile Chat			No
443	Static TCP	HTTPS reverse proxy				No
443	Static TCP	HTTPS STUN (ICE) Traffic				No
443	Static TCP	Access Edge SIP / TLS signaling				No
443	Static TCP	A / V Edge RTP / SRTP Media				No
6000 - 6006	TCP & UDP	Librestream Endpoints
10000-16000	TCP	H.245 control channel	No		No
10000-28000	UDP	RTP / SRTP Media	No		No
14085-15084	TCP	Edgewater / VBP H.225 / 245	No
16386-20385	UDP	Edgewater / VBP RTP Media	No
35061	TCP	Cloud App Alarm			No
30000-50000	TCP & UDP	Customer A / V Media			No
49152-49239	UDP	Sony Endpoints	No

Messaging

The H323 protocol performs messaging between two terminals and determines who will be the master and who will beslave, as well as the bandwidth of the participants and the audio and video codecs to be used. Terminals initiate the exchange of data, audio or video through the RTP / RTCP protocol. During the disconnect phase, any active link member can initiate the call termination process with Close Logical Channel and End Session Command messages, after which the RELEASE connection is closed.

The standard was originally developed to enable audio, video, and sharing communications between hosts connected to a corporate LAN and remote devices in the traditional PSTN circuit-switched network. It implements the H323 SIP Gateway Controller, responsible for user authentication and location, tracking registered clients.

In the process, the gatekeeper proxy communicates with the client, which reduces the load on low power client devices. Each terminal is identified by a pair (IP address, TCP/UDP port), so it can be directly contacted through its address/port pair without the use of a gatekeeper. If there is a gatekeeper, address/port pairs can be mapped to aliases so that users can remember them better, such as [email protected] Since they are associated with user accounts, aliases allow nomadism - the user will remain available even if they move by changing their IP address.

H323 call steps

The call takes place in several main steps:

1. Register - calls the terminal, searches for the gatekeeper in his area and opens the RAS channel,using its control.
2. Call Setup - The caller's terminal establishes a channel for the called party's terminal using call control.
3. Negotiation - Parameters such as bandwidth and codecs are negotiated using the control.
4. Data transmission - voice is transmitted by RTP close, data channel is closed by control.
5. End - the RAS channel is closed with the control.

The gatekeeper can play two roles:

1. The routed call goes through the gatekeeper, which is useful for NAT traversal, so the gatekeeper acts as a relay server.
2. Gatekeeper Direct Endpoint - The call is routed directly to the endpoint, but the caller and called party clients must first complete the Admission step for charging and bandwidth management.

Global protocol and modification differences

A "Session Initiation Protocol" (SIP) protocol standard developed by the IETF MMUSIC working group for initiating, modifying, and terminating user sessions: video, voice, and instant messaging.

The syntax of its operations resembles that of HTTP and SMTP, the protocols used in web page and email distribution services. This similarity is natural because SIP was designed to make telephony just another service on the Internet. This is newa standard for establishing, routing, and modifying communications over Internet Protocol (IP) networks. It takes the model of the internet and transforms it into a telecommunications world using existing internet protocols such as HTTP and SMTP (Simple Mail Transfer Protocol).

It also uses a URL structure to identify users instead of devices. Thus, SIP is device independent and does not distinguish between voice and data, phone or PC. SIP is more used for service management, while H323 performs the functions of converting the telephone standard into IP packets.

H323 was introduced as an evolution of SS7 designed for circuit switched signaling control. On the contrary, SIP is closer to HTTP, the packet network paradigm used on the Internet. Looking ahead, it is better to choose SIP. In this case, the media streams are transmitted using RTP, so the choice of control protocol or another does not directly affect the quality of the services offered. H 323 is much more complicated than SIP. It has hundreds of different messages encoded in binary format. Therefore, the H323 keeps developers as well as network administrators busy when troubleshooting.

Interoperability Scenarios

These protocols are widely used, so interoperability between SIP and H323 is essential to ensure full end-to-end connectivity. Due to inherent differences between H323 and SIP, consistency must be ensured in order to be able to interoperate. If they are used in the same administrative domain,call setup messages must be translated and then RTP can be used to communicate between the SIP phone and the H323 phone.

The scenario becomes more complex when the IP and H 323 gateway operate in separate administrative domains. This needs to be interpreted with a different protocol. H323 defines conferencing as part of the standard, including both centralized and decentralized. SIP does not have a definition for conferencing, but does have a process for conferencing that is similar to H323 but has not been formally defined as part of the standard.

Working with the Cisco C90 codec

Building a network H323:

1. First, codecs are configured based on IP addresses (192.168.2. XXX), correct subnet mask (255.255.255.0), gateway (192.168. XXX) - pfSense IP interface and DNS server. On C90 this is in admin settings -> IP setting.
2. NAT version for H323 is in Admin Settings -> Advanced -> H323 -> NAT.
3. Set: Mode=On.
4. Make sure the NAT address=WAN address so that the code can send the correct packets to the original connection.
5. Instruct the codec to disable connection through the gatekeeper. Under Administrator Settings -> Advanced -> H323 -> Profile 1 -> CallSetup.
6. Set: Mode=Direct.
7. H323 uses static ports for connections. Administrator Settings -> Advanced -> H323-> Profile 1.
8. Set: Port Allocation=Static.
9. Recheck H323 IP setting -> Optional -> Network 1 andmake sure the IP settings match the form.
10. Check the RTP ports to be used. The RTP stream transmits and receives audio and video. Under administrator settings -> Advanced -> RTP -> Ports -> Range. Pay attention to the "Start and Stop" values, you need to add them to the port firewall forwarding rules. The default is usually 2326 /2486.
11. Restart the codec for the settings to take effect.

After adding these rules, you can receive an H323 call from the codec. If the calls are not going through, you need to look in the logs to see if certain ports are being requested. You may need to configure H323 port forwarding settings and rules.

Benefits of choosing IP-telephony

With VoIP, you can make calls from anywhere you have a network connection. Since IP phones transmit information over the network, they are managed by providers from anywhere on the connection. This is an advantage for people who usually travel a lot and can carry their phone with them while accessing the VoIP service.

The benefits of VOIP services include:

1. Call identification.
2. Call waiting service.
3. Redirect service.
4. Recall.
5. Callback.
6. 3 line call (three way call).
7. Forward a call to a specific phone.

Obviously, both voice and video standards have their advantages and disadvantages, so instead of focusing onone standard over another, it makes more sense to work on improving the ways in which standards can be interoperated. This will provide end-to-end connectivity throughout the network, provide additional IP-oriented services. It is this modern approach that will demonstrate the full power of IP communications.